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webrtc_m130/common_audio
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Alessio Bazzica a33c7af42b Tolerate optional chunks in WAV files 
Wave files may contain optional chunks, such as a metadata one.
This CL makes WavReader tolerant to such chunks - it just ignores them.
For more details on the Wave format, please refer to
https://sites.google.com/site/musicgapi/technical-documents/wav-file-format.

Bug: webrtc:8762
Change-Id: Ie0e19dea75661808e7781f51faa1d0f0affeb3e1
Reviewed-on: https://webrtc-review.googlesource.com/c/40300
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25562}
2018-11-08 14:34:20 +00:00
..
include
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +00:00
mocks
Replace rtc::Optional with absl::optional
2018-06-21 09:32:56 +00:00
resampler
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +00:00
signal_processing
Remove simple stringstream usages.
2018-09-06 12:53:19 +00:00
third_party
Move spl_sqrt_floor dep to proper third_party directory
2018-08-07 10:33:13 +00:00
vad
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +00:00
audio_converter_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
audio_converter.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
audio_converter.h
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +00:00
audio_util_unittest.cc
Delete root header file typedef.h.
2018-07-25 14:59:26 +00:00
audio_util.cc
Delete root header file typedef.h.
2018-07-25 14:59:26 +00:00
BUILD.gn
Tolerate optional chunks in WAV files
2018-11-08 14:34:20 +00:00
channel_buffer_unittest.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
channel_buffer.cc
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +00:00
channel_buffer.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
DEPS
Removing openmax_dl
2018-05-07 12:12:04 +00:00
fir_filter_c.cc
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +00:00
fir_filter_c.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
fir_filter_factory.cc
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +00:00
fir_filter_factory.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
fir_filter_neon.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
fir_filter_neon.h
Move aligned memory utilities to rtc_base/memory/
2018-03-22 14:13:24 +00:00
fir_filter_sse.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
fir_filter_sse.h
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +00:00
fir_filter_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
fir_filter.h
Fix circular deps in common_audio.
2017-11-17 11:20:17 +00:00
OWNERS
…
real_fourier_ooura.cc
Move fft4g to proper third_party directory
2018-07-25 15:44:53 +00:00
real_fourier_ooura.h
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +00:00
real_fourier_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
real_fourier.cc
Removing warning suppression flags in common_audio/.
2018-07-04 11:13:09 +00:00
real_fourier.h
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +00:00
ring_buffer_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
ring_buffer.c
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
ring_buffer.h
Moving LappedTransform, Blocker and AudioRingBuffer.
2018-08-31 15:27:50 +00:00
smoothing_filter_unittest.cc
Replacing rtc::TimeDelta with webrtc::TimeDelta.
2018-05-08 13:22:53 +00:00
smoothing_filter.cc
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +00:00
smoothing_filter.h
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +00:00
sparse_fir_filter_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
sparse_fir_filter.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
sparse_fir_filter.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
wav_file_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
wav_file.cc
Tolerate optional chunks in WAV files
2018-11-08 14:34:20 +00:00
wav_file.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
wav_header_unittest.cc
Tolerate optional chunks in WAV files
2018-11-08 14:34:20 +00:00
wav_header.cc
Tolerate optional chunks in WAV files
2018-11-08 14:34:20 +00:00
wav_header.h
Tolerate optional chunks in WAV files
2018-11-08 14:34:20 +00:00
window_generator_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
window_generator.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
window_generator.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00